Tilghman Lesher [Thu, 10 Jan 2008 00:12:35 +0000 (00:12 +0000)]
Several manager changes:
1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
Terry Wilson [Wed, 9 Jan 2008 21:37:26 +0000 (21:37 +0000)]
Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
Luigi Rizzo [Wed, 9 Jan 2008 18:03:40 +0000 (18:03 +0000)]
Two changes:
- support scrolling of message window;
- simplify the code for creating a message window,
and try it using a second one in the top of
the keypad (where we echo the dialed number).
The 'skin' that supports these two windows will be
committed separately.
Set the caller id within the gtalk_alloc function.
As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Don't do conferencing totally in Zaptel if Monitor is running on the channel.
(closes issue #11709)
Reported by: BigJimmy
Patches:
patch-meetmerec uploaded by BigJimmy (license 371)
Luigi Rizzo [Wed, 9 Jan 2008 12:23:18 +0000 (12:23 +0000)]
implement the "console startgui" and "console stopgui"
commands so you can start and stop the gui even outside
of a call. This is convenient for testing, and also for
using the keypad to pick up a call, and to dial a number
(the latter not yet implemented, but should be close).
Allow filename completion on zero-length modules, remove a memory leak, remove
a file descriptor leak, and make filename completion thread-safe.
Patched and tested by tilghman.
(Closes issue #11681)
Part 1 of N of adding doxygen comments to app_queue. I picked some of the most common functions
used (which also happen to be some the biggest/ugliest functions too) to document first. I'm pretty
new to doxygen so criticism is welcome.
Tilghman Lesher [Tue, 8 Jan 2008 23:51:51 +0000 (23:51 +0000)]
Add a new flag 'd' (with optional context) permitting any extension within
that context to be entered as a new extension during the playback of a
voicemail greeting.
Patch inspired by bluecrow76, by tilghman.
(Closes issue #7063)
Luigi Rizzo [Tue, 8 Jan 2008 23:14:48 +0000 (23:14 +0000)]
Add the annotation for the textarea used for messages,
and also change the background from white to something
different to show that we can make use of fonts with transparent
background.
Luigi Rizzo [Tue, 8 Jan 2008 23:11:25 +0000 (23:11 +0000)]
add a font suitable for use with the console GUI.
The background of this particular image is transparent so we
can preserve the original background when we draw strings.
Luigi Rizzo [Tue, 8 Jan 2008 23:09:44 +0000 (23:09 +0000)]
add support for textareas, used for various dialog windows on the gui.
The main code to implement the textarea is in console_board.c,
and uses a simple png image with the font, blitting characters
on the designated areas of the main screen.
Additionally we provide some annotations in the image used
as a skin to indicate which areas are used for text messages.
(images will be committed separately).
At the moment the dialog area is only used to display a running
counter, just as a proof of concept.
Mark Michelson [Tue, 8 Jan 2008 21:18:32 +0000 (21:18 +0000)]
Adding the option of specifying a second interface in a member definition for a queue. app_queue
will monitor this second device's state for the member, even though it actually calls the first
interface. This ability has been added for statically defined queue members, realtime queue members,
and dynamic queue members added through the CLI, dialplan, or manager.
Making some changes designed to not allow for a corrupted mailstream for a vm_state.
1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs.
2. Make sure to always grab the persistent vm_state when mailstream access is necessary.
3. Correct an incorrect return value in the init_mailstream function.
Joshua Colp [Tue, 8 Jan 2008 20:06:52 +0000 (20:06 +0000)]
Move common code for setting T38 capabilities and fix a bug with fax detection in the SIP RTP read callback. It's still sort of silly... but more on that later.
(closes issue #11239)
Reported by: dimas
Patches:
sipt38prop.patch uploaded by dimas (license 88)
If no group has been provided to the GROUP_COUNT dialplan function then use the first one specific to the channel.
(closes issue #11077)
Reported by: m4him
Luigi Rizzo [Mon, 7 Jan 2008 23:03:11 +0000 (23:03 +0000)]
add support for cropping the keypad image while displaying it.
This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
Mark Michelson [Mon, 7 Jan 2008 21:04:09 +0000 (21:04 +0000)]
Adding user-configurable TCP timeout settings to IMAP voicemail. This could
go a long way towards preventing unexplainable hangs experienced by people. In the
case of MWI hangs, this also will mean that the SIP port isn't blocked anymore.
Russell Bryant [Mon, 7 Jan 2008 17:09:23 +0000 (17:09 +0000)]
If the HTML documentation exists, install it in the static-http/docs directory
so that it can be viewed through the Asterisk http server if it is turned on.
Joshua Colp [Mon, 7 Jan 2008 15:52:55 +0000 (15:52 +0000)]
Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. Previously they would get registered twice because of the way manager.c operates.
(closes issue #11699)
Reported by: caio1982
Patches:
manager_module_commands1.diff uploaded by caio1982 (license 22)
Russell Bryant [Sat, 5 Jan 2008 23:05:35 +0000 (23:05 +0000)]
Convert this file over the new method of getting the Asterisk version.
(I don't have this building on this machine, so caio1982 on IRC is going to
test it for me. :) )
Russell Bryant [Sat, 5 Jan 2008 22:09:06 +0000 (22:09 +0000)]
Now that the version.h file was getting properly regenerated every time the svn
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
Russell Bryant [Sat, 5 Jan 2008 21:35:54 +0000 (21:35 +0000)]
Add a note about changing modules.conf since another console channel driver is
now present that can not be used at the same time as chan_alsa or chan_oss.
Fix the problem of notification of a device state change to a device with a '-'
in the name. Could probably do with a better fix in trunk, but this bug has
been open way too long without a better solution.
Reported by: stevedavies
Patch by: tilghman
(Closes issue #9668)
Philippe Sultan [Fri, 4 Jan 2008 17:19:25 +0000 (17:19 +0000)]
Use SASL DIGEST-MD5 authentication over unsecured network connections only.
This authentication mechanism is implemented under the iksemel API, which
makes use of GnuTLS, whereas we use OpenSSL. Note : there's ongoing
dicsussion at the SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
http://ietfreport.isoc.org/ids-wg-sasl.html.
eliminiate sound_thread() and other stuff from chan_oss since Asterisk indications can handle it
remove gentone and all the headers containing tones that are no longer needed
coding guidelines cleanup
remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
Merged revisions 96198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line
when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense.
........
We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that
multiple members can have the same name, since the variable was not reset on each iteration of the loop.
Russell Bryant [Wed, 2 Jan 2008 23:22:25 +0000 (23:22 +0000)]
Add support for generating a ringing sound on an incoming call. This is a bit
of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl if it is present, but doesn't parse any supplied parameters yet
(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)
Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001)
(closes issue #11637)
Reported by: greyvoip
projects / thirdparty / asterisk.git / log
